Legal claims defining the scope of protection. Each claim is shown in both the original legal language and a plain English translation.
1. A method for processing an audio signal, the method comprising: receiving an input audio signal; receiving a set of truncated subband filter coefficients for each subband and each channel, wherein the set of truncated subband filter coefficients is truncated frequency dependently from a set of subband filter coefficients of a binaural room impulse response (BRIR) data set, wherein the length of the set of truncated subband filter coefficients is determined based on a filter order of the corresponding subband, and wherein the filter order is determined to be variable in the frequency domain; obtaining vector information indicating a particular BRIR data set corresponding to a relevant channel of the input audio signal; and filtering each subband signal of the input audio signal by using the set of truncated subband filter coefficients corresponding to the relevant channel and a relevant subband based on the vector information.
Audio signal processing, specifically for simulating spatial audio effects. The problem addressed is efficiently applying binaural room impulse response (BRIR) data to an audio signal while managing computational complexity. The invention involves a method for processing an audio signal. An input audio signal is received. A set of truncated subband filter coefficients is also received. These coefficients are generated for each subband and each audio channel. Crucially, the coefficients are truncated frequency dependently from a larger set of subband filter coefficients derived from a BRIR data set. The length of these truncated coefficients is determined by a filter order, which itself varies across the frequency domain for each subband. Additionally, vector information is obtained. This information identifies a specific BRIR data set relevant to a particular channel of the input audio signal. Finally, each subband signal of the input audio signal is filtered. This filtering process utilizes the previously received set of truncated subband filter coefficients. The selection of which coefficients to use is based on the relevant channel identified by the vector information and the relevant subband. This approach allows for a computationally efficient application of BRIR data by using frequency-dependent, variable-order truncated filter coefficients.
2. The method of claim 1 , wherein when a first BRIR data set having positional information matching with positional information of the relevant channel of the input audio signal is present in a predetermined BRIR filter set, the vector information indicates the first BRIR data set as the particular BRIR data set corresponding to the relevant channel.
This invention relates to audio signal processing, specifically methods for selecting binaural room impulse response (BRIR) data sets to process input audio signals for spatial audio rendering. The problem addressed is efficiently matching input audio channels with appropriate BRIR data sets to accurately reproduce spatial audio effects, such as those used in virtual reality or 3D audio systems. The method involves analyzing an input audio signal containing multiple channels, each representing a sound source in a virtual space. For each channel, the system checks if a BRIR data set exists in a predefined filter set that has positional information matching the channel's positional information. If a matching BRIR data set is found, the system selects that data set for processing the corresponding audio channel. The BRIR data set contains impulse response information that models how sound propagates from the source position to a listener's ears, enabling realistic spatial audio reproduction. The method ensures that each audio channel is processed with the most accurate BRIR data available, improving the realism of the spatial audio experience. This approach is particularly useful in applications where precise positional audio rendering is critical, such as virtual reality environments or immersive audio systems. The system dynamically selects the best-matching BRIR data set for each channel, optimizing both computational efficiency and audio quality.
3. The method of claim 1 , wherein when a first BRIR data set having positional information matching with positional information of the relevant channel of the input audio signal is not present in a predetermined BRIR filter set, the vector information indicates a second BRIR data set having a minimum geometric distance from the positional information of the relevant channel as the particular BRIR data set corresponding to the relevant channel.
This invention relates to audio signal processing, specifically methods for selecting binaural room impulse response (BRIR) data sets to process input audio signals for spatial audio rendering. The problem addressed is efficiently matching input audio channels to appropriate BRIR data sets when an exact positional match is unavailable in a predefined BRIR filter set. The method involves analyzing an input audio signal containing multiple channels, each with associated positional information. For each channel, the system checks if a BRIR data set with matching positional information exists in a predetermined BRIR filter set. If no exact match is found, the system identifies a second BRIR data set from the filter set that has the minimum geometric distance from the channel's positional information. This second BRIR data set is then selected as the closest available match for processing the relevant channel. The geometric distance calculation ensures that the selected BRIR data set is the most positionally similar alternative when an exact match is absent, optimizing spatial audio rendering accuracy. This approach improves the flexibility and robustness of spatial audio systems by dynamically adapting to available BRIR data sets while maintaining positional relevance.
4. The method of claim 3 , wherein the geometric distance is a value obtained by aggregating an absolute value of an altitude deviation between two positions and an absolute value of an azimuth deviation between the two positions.
This invention relates to a method for determining a geometric distance between two positions in a three-dimensional space, particularly for applications in navigation, robotics, or spatial analysis. The problem addressed is the need for an accurate and computationally efficient way to measure the spatial separation between two points, accounting for both vertical (altitude) and horizontal (azimuth) deviations. The method calculates the geometric distance by aggregating two components: the absolute value of the altitude deviation between the two positions and the absolute value of the azimuth deviation between the two positions. The altitude deviation represents the vertical difference in elevation or height between the two points, while the azimuth deviation represents the horizontal angular difference or lateral displacement. By combining these two measurements, the method provides a comprehensive metric that reflects both vertical and horizontal separation. This approach is particularly useful in scenarios where traditional Euclidean distance calculations may not fully capture the spatial relationship between points, such as in uneven terrain or three-dimensional environments. The aggregation of altitude and azimuth deviations ensures that the geometric distance is sensitive to both vertical and horizontal changes, making it suitable for applications requiring precise spatial measurements. The method can be implemented in systems where accurate positioning or path planning is critical, such as autonomous vehicles, drones, or surveying tools.
5. The method of claim 1 , wherein a length of the set of truncated subband filter coefficients of at least one subband is different from a length of the set of truncated subband filter coefficients of another subband.
This invention relates to digital signal processing, specifically to methods for optimizing subband filter coefficients in filter bank systems. The problem addressed is the inefficiency in traditional filter banks where subband filters use uniform coefficient lengths, leading to unnecessary computational overhead and memory usage. The solution involves truncating subband filter coefficients to different lengths for different subbands, allowing for a more efficient implementation while maintaining signal processing performance. The method processes an input signal by decomposing it into multiple subbands using a filter bank. Each subband is filtered using a set of subband filter coefficients, but unlike conventional systems, the length of these coefficient sets varies between subbands. For example, subbands with less critical frequency components may use shorter coefficient sets, while subbands with more critical components retain longer sets. This selective truncation reduces computational complexity and memory requirements without degrading signal quality. The invention also includes determining the optimal truncation points for each subband based on factors such as signal characteristics, desired performance, and hardware constraints. The truncated coefficients are then applied in the filter bank, ensuring efficient processing while preserving the integrity of the filtered signal. This approach is particularly useful in applications like audio processing, wireless communications, and multimedia systems where computational efficiency is critical.
6. An apparatus for processing an audio signal for performing binaural rendering for an input audio signal, the apparatus comprising: a binaural rendering unit configured to: receive an input audio signal, receive a set of truncated subband filter coefficients for each subband and each channel, wherein the set of truncated subband filter coefficients is truncated frequency dependently from a set of subband filter coefficients of a binaural room impulse response (BRIR) data set, wherein the length of the set of truncated subband filter coefficients is determined based on a filter order of the corresponding subband, and wherein the filter order is determined to be variable in the frequency domain, obtain vector information indicating a particular BRIR data set corresponding to a relevant channel of the input audio signal, and filter each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and a relevant subband based on the vector information.
This invention relates to audio signal processing for binaural rendering, which aims to simulate three-dimensional sound perception using headphones. The problem addressed is the computational complexity and memory requirements of traditional binaural rendering systems, which use full-length binaural room impulse responses (BRIRs) for each subband and channel. These full-length BRIRs are resource-intensive, especially for high-frequency subbands where shorter filter lengths are sufficient. The apparatus processes an input audio signal by applying binaural rendering through a specialized unit. The unit receives the input audio signal and a set of truncated subband filter coefficients for each subband and channel. These coefficients are derived from a BRIR data set but are truncated in a frequency-dependent manner. The truncation length is determined by the filter order of the corresponding subband, which varies across the frequency domain. Higher frequencies use shorter filters, reducing computational load while maintaining audio quality. The unit also receives vector information that identifies the relevant BRIR data set for the input signal's channel. The subband signals are then filtered using the truncated coefficients specific to the channel and subband, as indicated by the vector information. This approach optimizes processing efficiency by adapting filter lengths to the frequency characteristics of the audio signal.
7. The apparatus of claim 6 , wherein when a first BRIR data set having positional information matching with positional information of the relevant channel of the input audio signal is present in a predetermined BRIR filter set, the vector information indicates the first BRIR data set as the particular BRIR data set corresponding to the relevant channel.
This invention relates to audio processing systems, specifically methods for selecting binaural room impulse response (BRIR) data sets for spatial audio rendering. The problem addressed is efficiently matching input audio signals to appropriate BRIR filters to accurately reproduce spatial audio characteristics, such as directionality and room acoustics, in a headphone or speaker system. The system processes an input audio signal containing multiple channels, each representing a different sound source position. For each channel, the system determines positional information indicating the source's location in a 3D space. The system then searches a predetermined set of BRIR filters, each associated with specific positional data, to find a matching BRIR data set. When a first BRIR data set is found with positional information that matches the channel's positional information, the system selects this data set for processing the channel. The selection is indicated by vector information that points to the matching BRIR data set, ensuring the correct spatial characteristics are applied to the audio signal. This approach improves audio rendering accuracy by dynamically selecting the most appropriate BRIR filters based on real-time positional data, enhancing the realism of spatial audio reproduction.
8. The apparatus of claim 6 , wherein when a first BRIR data set having positional information matching with positional information of the relevant channel of the input audio signal is not present in a predetermined BRIR filter set, the vector information indicates a second BRIR data set having a minimum geometric distance from the positional information of the relevant channel as the particular BRIR data set corresponding to the relevant channel.
This invention relates to audio processing systems that use Binaural Room Impulse Response (BRIR) filters to simulate spatial audio. The problem addressed is efficiently selecting the most appropriate BRIR data set for processing input audio signals when an exact positional match is unavailable. When a first BRIR data set with positional information matching the relevant channel of the input audio signal is missing from a predetermined BRIR filter set, the system identifies a second BRIR data set with the minimum geometric distance from the target position. This ensures the closest available spatial approximation is used, maintaining audio quality. The apparatus includes a BRIR filter set containing multiple BRIR data sets, each with associated positional information. A vector information generator evaluates the positional data and selects the nearest BRIR data set when an exact match is absent. This approach optimizes real-time audio rendering by minimizing computational overhead while preserving spatial accuracy. The solution is particularly useful in virtual reality, augmented reality, and immersive audio applications where precise spatial audio reproduction is critical.
9. The apparatus of claim 8 , wherein the geometric distance is a value obtained by aggregating an absolute value of an altitude deviation between two positions and an absolute value of an azimuth deviation between the two positions.
This invention relates to a system for determining a geometric distance between two positions in a three-dimensional space, particularly for applications in navigation, robotics, or positioning systems. The problem addressed is the need for an accurate and computationally efficient method to calculate spatial separation between two points, accounting for both vertical (altitude) and horizontal (azimuth) deviations. The apparatus includes a processing unit configured to compute the geometric distance by aggregating two components: the absolute value of the altitude deviation between the two positions and the absolute value of the azimuth deviation between the two positions. The altitude deviation represents the vertical difference in elevation, while the azimuth deviation represents the horizontal angular difference. By combining these two measurements, the system provides a comprehensive metric of spatial separation that considers both vertical and horizontal displacement. This approach improves upon traditional distance calculations by incorporating directional and elevation differences, which is particularly useful in environments where terrain or obstacles require precise positional awareness. The method ensures that the geometric distance accurately reflects the true spatial relationship between the two points, enhancing navigation accuracy and reliability in dynamic or complex environments. The system may be integrated into devices such as drones, autonomous vehicles, or surveying tools to improve positional tracking and decision-making.
10. The apparatus of claim 6 , wherein a length of the set of truncated subband filter coefficients of at least one subband is different from a length of the set of truncated subband filter coefficients of another subband.
This invention relates to digital signal processing, specifically to subband filtering techniques used in audio or communication systems. The problem addressed is the inefficiency of conventional subband filters, which often use uniform filter lengths across all subbands, leading to suboptimal performance in terms of computational complexity and signal quality. The apparatus includes a subband filter system that processes input signals by dividing them into multiple subbands. Each subband is filtered using a set of truncated subband filter coefficients. A key feature is that the length of the truncated filter coefficients can vary between subbands. For example, one subband may use a shorter filter length to reduce computational load, while another subband may use a longer filter length to improve signal fidelity. This adaptability allows for optimized performance based on the characteristics of each subband, such as frequency content or signal importance. The system may also include a coefficient truncation module that determines the appropriate length for each subband's filter coefficients based on predefined criteria, such as signal-to-noise ratio requirements or computational constraints. The truncated coefficients are then applied to the corresponding subband signals, resulting in an output signal with improved efficiency and quality. This approach is particularly useful in applications where different subbands require different levels of precision, such as audio coding, speech enhancement, or wireless communication systems.
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September 8, 2020
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